Call gets disconnected when another call comes in

General Help
1

Hey,
One of my client is experiencing an issue where the call gets disconnected as soon as another call comes in. But its quite intermittent.

Need urgent help please

2

You provided no information on your DAHDI version, how the system was installed (disto or by hand) and what version of FreePBX.

Also the log /var/log/asterisk should provide clues if you can catch the event occurring.

3

Asterisk (Ver. 1.6.0.26)

4

When you mean disto or by I think by hand. I am not an expert on asterisk sorry

5

Did you install from an ISO distribution or did you download it and install it on an existing Linux system? If ISO, which one?

Also if you do not know Asterisk and FreePBX why are you doing this professionally for clients?

6

Its an ISO and fresh installation.
I know the difference
I said I am not an expert

7

OK, so answer the rest of my question.

8

dahdi-tools-2.3.0-1_trixbox
dahdi-tools-doc-2.2.0-4_trixbox
dahdi-linux-2.3.0.1-1_trixbox
asterisk16-dahdi-1.6.0.26-1_trixbox
kmod-dahdi-linux-2.3.0.1-1_trixbox.2.6.18_164.11.1.el5

9

I think I’d like to know what type trunks are being used.

10

You installed trixbox? Why would you do that?

Did you notice my (and other folks) warnings all over the site that the project is abandoned? Long story but Fonality walked away and left all the users hanging.

The trixbox DAHDI and Asterisk is all hacked up. You probably would not have this issue with a current version of Asterisk/FreePBX

11

Hey it is an old implementation and upgrading or changing can be a
Mess. Any suggestions as to what be causing even though it is not supported.
It will be greatly appreciated if you can help me resolve this issue.

It’s SIP trunks and we got the sip providers to investigate and we noticed another disconnects the present call and it’s intermittent

12

You said it was a “fresh install”

There are too many bugs and no documentation on the changes Fonality made to Asterisk and FreePBX to make support practical.

It’s not that hard to convert the systems, especially if you know how to use vmplayer on a PC to test the conversion.

If you just can’t update I would put a sniffer on the LAN and check the logs and see if you have a correlating event. Ethereal does a nice job of decoding SIP messages.

13

I meant when it was installed it was a fresh installation my bad, gave you the wrong impression